A network performance comparison of webrtc and sip audio and video communications

Multimedia services are one of the internetneeds with high data traffic network count since 2019. Twoof the multimedia services, Web Real-TimeCommunications (WebRTC) and Session InitiationProtocol (SIP), have been widely used in applications forconducting video conferencing. The main objective of thisresearch is to analyze network performance by anapplication with a client-server for audio and videocommunications developed with WebRTC and SIPprotocols. The SIP system uses the FreePBX server, andthe softphone application uses Bria. Whereas WebRTCuses JavaScript with servers on Ubuntu using Node.js. Theanalysis application uses a star topology and runs on alocal network using Wi-Fi. After testing, the results showthat the throughput, jitter, and packet loss of WebRTC arebetter than SIP. This result is caused by several factorsincluding the type of codec, the type of platform, and theway of signaling and routing of each protocol.